47d0a9264f
Add API to retrieve number of frames dropped by audio input kernel driver. Submitted on behalf of Masaki Sato <masaki.sato@motorola.com>
932 lines
28 KiB
C++
932 lines
28 KiB
C++
/*
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* Copyright (C) 2006-2007 The Android Open Source Project
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*
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* Licensed under the Apache License, Version 2.0 (the "License");
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* you may not use this file except in compliance with the License.
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* You may obtain a copy of the License at
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*
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* http://www.apache.org/licenses/LICENSE-2.0
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*
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* Unless required by applicable law or agreed to in writing, software
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* distributed under the License is distributed on an "AS IS" BASIS,
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* WITHOUT WARRANTIES OR CONDITIONS OF ANY KIND, either express or implied.
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* See the License for the specific language governing permissions and
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* limitations under the License.
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*/
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#define LOG_TAG "AudioSystem"
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//#define LOG_NDEBUG 0
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#include <utils/Log.h>
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#include <binder/IServiceManager.h>
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#include <media/AudioSystem.h>
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#include <media/IAudioPolicyService.h>
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#include <math.h>
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// ----------------------------------------------------------------------------
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// the sim build doesn't have gettid
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#ifndef HAVE_GETTID
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# define gettid getpid
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#endif
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// ----------------------------------------------------------------------------
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namespace android {
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// client singleton for AudioFlinger binder interface
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Mutex AudioSystem::gLock;
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sp<IAudioFlinger> AudioSystem::gAudioFlinger;
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sp<AudioSystem::AudioFlingerClient> AudioSystem::gAudioFlingerClient;
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audio_error_callback AudioSystem::gAudioErrorCallback = NULL;
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// Cached values
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DefaultKeyedVector<int, audio_io_handle_t> AudioSystem::gStreamOutputMap(0);
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DefaultKeyedVector<audio_io_handle_t, AudioSystem::OutputDescriptor *> AudioSystem::gOutputs(0);
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// Cached values for recording queries
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uint32_t AudioSystem::gPrevInSamplingRate = 16000;
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int AudioSystem::gPrevInFormat = AudioSystem::PCM_16_BIT;
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int AudioSystem::gPrevInChannelCount = 1;
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size_t AudioSystem::gInBuffSize = 0;
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// establish binder interface to AudioFlinger service
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const sp<IAudioFlinger>& AudioSystem::get_audio_flinger()
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{
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Mutex::Autolock _l(gLock);
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if (gAudioFlinger.get() == 0) {
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sp<IServiceManager> sm = defaultServiceManager();
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sp<IBinder> binder;
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do {
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binder = sm->getService(String16("media.audio_flinger"));
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if (binder != 0)
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break;
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LOGW("AudioFlinger not published, waiting...");
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usleep(500000); // 0.5 s
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} while(true);
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if (gAudioFlingerClient == NULL) {
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gAudioFlingerClient = new AudioFlingerClient();
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} else {
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if (gAudioErrorCallback) {
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gAudioErrorCallback(NO_ERROR);
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}
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}
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binder->linkToDeath(gAudioFlingerClient);
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gAudioFlinger = interface_cast<IAudioFlinger>(binder);
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gAudioFlinger->registerClient(gAudioFlingerClient);
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}
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LOGE_IF(gAudioFlinger==0, "no AudioFlinger!?");
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return gAudioFlinger;
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}
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status_t AudioSystem::muteMicrophone(bool state) {
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const sp<IAudioFlinger>& af = AudioSystem::get_audio_flinger();
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if (af == 0) return PERMISSION_DENIED;
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return af->setMicMute(state);
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}
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status_t AudioSystem::isMicrophoneMuted(bool* state) {
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const sp<IAudioFlinger>& af = AudioSystem::get_audio_flinger();
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if (af == 0) return PERMISSION_DENIED;
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*state = af->getMicMute();
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return NO_ERROR;
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}
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status_t AudioSystem::setMasterVolume(float value)
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{
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const sp<IAudioFlinger>& af = AudioSystem::get_audio_flinger();
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if (af == 0) return PERMISSION_DENIED;
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af->setMasterVolume(value);
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return NO_ERROR;
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}
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status_t AudioSystem::setMasterMute(bool mute)
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{
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const sp<IAudioFlinger>& af = AudioSystem::get_audio_flinger();
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if (af == 0) return PERMISSION_DENIED;
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af->setMasterMute(mute);
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return NO_ERROR;
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}
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status_t AudioSystem::getMasterVolume(float* volume)
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{
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const sp<IAudioFlinger>& af = AudioSystem::get_audio_flinger();
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if (af == 0) return PERMISSION_DENIED;
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*volume = af->masterVolume();
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return NO_ERROR;
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}
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status_t AudioSystem::getMasterMute(bool* mute)
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{
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const sp<IAudioFlinger>& af = AudioSystem::get_audio_flinger();
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if (af == 0) return PERMISSION_DENIED;
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*mute = af->masterMute();
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return NO_ERROR;
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}
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status_t AudioSystem::setStreamVolume(int stream, float value, int output)
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{
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if (uint32_t(stream) >= NUM_STREAM_TYPES) return BAD_VALUE;
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const sp<IAudioFlinger>& af = AudioSystem::get_audio_flinger();
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if (af == 0) return PERMISSION_DENIED;
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af->setStreamVolume(stream, value, output);
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return NO_ERROR;
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}
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status_t AudioSystem::setStreamMute(int stream, bool mute)
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{
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if (uint32_t(stream) >= NUM_STREAM_TYPES) return BAD_VALUE;
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const sp<IAudioFlinger>& af = AudioSystem::get_audio_flinger();
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if (af == 0) return PERMISSION_DENIED;
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af->setStreamMute(stream, mute);
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return NO_ERROR;
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}
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status_t AudioSystem::getStreamVolume(int stream, float* volume, int output)
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{
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if (uint32_t(stream) >= NUM_STREAM_TYPES) return BAD_VALUE;
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const sp<IAudioFlinger>& af = AudioSystem::get_audio_flinger();
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if (af == 0) return PERMISSION_DENIED;
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*volume = af->streamVolume(stream, output);
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return NO_ERROR;
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}
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status_t AudioSystem::getStreamMute(int stream, bool* mute)
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{
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if (uint32_t(stream) >= NUM_STREAM_TYPES) return BAD_VALUE;
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const sp<IAudioFlinger>& af = AudioSystem::get_audio_flinger();
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if (af == 0) return PERMISSION_DENIED;
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*mute = af->streamMute(stream);
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return NO_ERROR;
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}
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status_t AudioSystem::setMode(int mode)
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{
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if (mode >= NUM_MODES) return BAD_VALUE;
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const sp<IAudioFlinger>& af = AudioSystem::get_audio_flinger();
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if (af == 0) return PERMISSION_DENIED;
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return af->setMode(mode);
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}
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status_t AudioSystem::isStreamActive(int stream, bool* state) {
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const sp<IAudioFlinger>& af = AudioSystem::get_audio_flinger();
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if (af == 0) return PERMISSION_DENIED;
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*state = af->isStreamActive(stream);
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return NO_ERROR;
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}
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status_t AudioSystem::setParameters(audio_io_handle_t ioHandle, const String8& keyValuePairs) {
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const sp<IAudioFlinger>& af = AudioSystem::get_audio_flinger();
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if (af == 0) return PERMISSION_DENIED;
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return af->setParameters(ioHandle, keyValuePairs);
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}
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String8 AudioSystem::getParameters(audio_io_handle_t ioHandle, const String8& keys) {
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const sp<IAudioFlinger>& af = AudioSystem::get_audio_flinger();
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String8 result = String8("");
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if (af == 0) return result;
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result = af->getParameters(ioHandle, keys);
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return result;
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}
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// convert volume steps to natural log scale
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// change this value to change volume scaling
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static const float dBPerStep = 0.5f;
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// shouldn't need to touch these
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static const float dBConvert = -dBPerStep * 2.302585093f / 20.0f;
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static const float dBConvertInverse = 1.0f / dBConvert;
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float AudioSystem::linearToLog(int volume)
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{
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// float v = volume ? exp(float(100 - volume) * dBConvert) : 0;
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// LOGD("linearToLog(%d)=%f", volume, v);
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// return v;
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return volume ? exp(float(100 - volume) * dBConvert) : 0;
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}
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int AudioSystem::logToLinear(float volume)
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{
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// int v = volume ? 100 - int(dBConvertInverse * log(volume) + 0.5) : 0;
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// LOGD("logTolinear(%d)=%f", v, volume);
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// return v;
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return volume ? 100 - int(dBConvertInverse * log(volume) + 0.5) : 0;
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}
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status_t AudioSystem::getOutputSamplingRate(int* samplingRate, int streamType)
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{
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OutputDescriptor *outputDesc;
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audio_io_handle_t output;
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if (streamType == DEFAULT) {
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streamType = MUSIC;
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}
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output = getOutput((stream_type)streamType);
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if (output == 0) {
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return PERMISSION_DENIED;
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}
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gLock.lock();
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outputDesc = AudioSystem::gOutputs.valueFor(output);
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if (outputDesc == 0) {
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LOGV("getOutputSamplingRate() no output descriptor for output %d in gOutputs", output);
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gLock.unlock();
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const sp<IAudioFlinger>& af = AudioSystem::get_audio_flinger();
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if (af == 0) return PERMISSION_DENIED;
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*samplingRate = af->sampleRate(output);
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} else {
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LOGV("getOutputSamplingRate() reading from output desc");
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*samplingRate = outputDesc->samplingRate;
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gLock.unlock();
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}
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LOGV("getOutputSamplingRate() streamType %d, output %d, sampling rate %d", streamType, output, *samplingRate);
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return NO_ERROR;
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}
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status_t AudioSystem::getOutputFrameCount(int* frameCount, int streamType)
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{
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OutputDescriptor *outputDesc;
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audio_io_handle_t output;
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if (streamType == DEFAULT) {
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streamType = MUSIC;
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}
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output = getOutput((stream_type)streamType);
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if (output == 0) {
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return PERMISSION_DENIED;
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}
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gLock.lock();
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outputDesc = AudioSystem::gOutputs.valueFor(output);
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if (outputDesc == 0) {
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gLock.unlock();
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const sp<IAudioFlinger>& af = AudioSystem::get_audio_flinger();
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if (af == 0) return PERMISSION_DENIED;
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*frameCount = af->frameCount(output);
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} else {
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*frameCount = outputDesc->frameCount;
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gLock.unlock();
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}
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LOGV("getOutputFrameCount() streamType %d, output %d, frameCount %d", streamType, output, *frameCount);
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return NO_ERROR;
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}
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status_t AudioSystem::getOutputLatency(uint32_t* latency, int streamType)
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{
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OutputDescriptor *outputDesc;
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audio_io_handle_t output;
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if (streamType == DEFAULT) {
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streamType = MUSIC;
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}
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output = getOutput((stream_type)streamType);
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if (output == 0) {
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return PERMISSION_DENIED;
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}
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gLock.lock();
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outputDesc = AudioSystem::gOutputs.valueFor(output);
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if (outputDesc == 0) {
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gLock.unlock();
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const sp<IAudioFlinger>& af = AudioSystem::get_audio_flinger();
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if (af == 0) return PERMISSION_DENIED;
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*latency = af->latency(output);
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} else {
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*latency = outputDesc->latency;
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gLock.unlock();
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}
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LOGV("getOutputLatency() streamType %d, output %d, latency %d", streamType, output, *latency);
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return NO_ERROR;
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}
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status_t AudioSystem::getInputBufferSize(uint32_t sampleRate, int format, int channelCount,
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size_t* buffSize)
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{
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// Do we have a stale gInBufferSize or are we requesting the input buffer size for new values
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if ((gInBuffSize == 0) || (sampleRate != gPrevInSamplingRate) || (format != gPrevInFormat)
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|| (channelCount != gPrevInChannelCount)) {
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// save the request params
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gPrevInSamplingRate = sampleRate;
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gPrevInFormat = format;
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gPrevInChannelCount = channelCount;
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gInBuffSize = 0;
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const sp<IAudioFlinger>& af = AudioSystem::get_audio_flinger();
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if (af == 0) {
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return PERMISSION_DENIED;
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}
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gInBuffSize = af->getInputBufferSize(sampleRate, format, channelCount);
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}
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*buffSize = gInBuffSize;
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return NO_ERROR;
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}
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status_t AudioSystem::setVoiceVolume(float value)
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{
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const sp<IAudioFlinger>& af = AudioSystem::get_audio_flinger();
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if (af == 0) return PERMISSION_DENIED;
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return af->setVoiceVolume(value);
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}
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status_t AudioSystem::getRenderPosition(uint32_t *halFrames, uint32_t *dspFrames, int stream)
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{
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const sp<IAudioFlinger>& af = AudioSystem::get_audio_flinger();
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if (af == 0) return PERMISSION_DENIED;
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if (stream == DEFAULT) {
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stream = MUSIC;
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}
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return af->getRenderPosition(halFrames, dspFrames, getOutput((stream_type)stream));
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}
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unsigned int AudioSystem::getInputFramesLost(audio_io_handle_t ioHandle) {
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const sp<IAudioFlinger>& af = AudioSystem::get_audio_flinger();
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unsigned int result = 0;
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if (af == 0) return result;
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if (ioHandle == NULL) return result;
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result = af->getInputFramesLost(ioHandle);
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return result;
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}
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// ---------------------------------------------------------------------------
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void AudioSystem::AudioFlingerClient::binderDied(const wp<IBinder>& who) {
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Mutex::Autolock _l(AudioSystem::gLock);
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AudioSystem::gAudioFlinger.clear();
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// clear output handles and stream to output map caches
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AudioSystem::gStreamOutputMap.clear();
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AudioSystem::gOutputs.clear();
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if (gAudioErrorCallback) {
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gAudioErrorCallback(DEAD_OBJECT);
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}
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LOGW("AudioFlinger server died!");
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}
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void AudioSystem::AudioFlingerClient::ioConfigChanged(int event, int ioHandle, void *param2) {
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LOGV("ioConfigChanged() event %d", event);
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OutputDescriptor *desc;
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uint32_t stream;
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if (ioHandle == 0) return;
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Mutex::Autolock _l(AudioSystem::gLock);
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switch (event) {
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case STREAM_CONFIG_CHANGED:
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if (param2 == 0) break;
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stream = *(uint32_t *)param2;
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LOGV("ioConfigChanged() STREAM_CONFIG_CHANGED stream %d, output %d", stream, ioHandle);
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if (gStreamOutputMap.indexOfKey(stream) >= 0) {
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gStreamOutputMap.replaceValueFor(stream, ioHandle);
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}
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break;
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case OUTPUT_OPENED: {
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if (gOutputs.indexOfKey(ioHandle) >= 0) {
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LOGV("ioConfigChanged() opening already existing output! %d", ioHandle);
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break;
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}
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if (param2 == 0) break;
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desc = (OutputDescriptor *)param2;
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OutputDescriptor *outputDesc = new OutputDescriptor(*desc);
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gOutputs.add(ioHandle, outputDesc);
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LOGV("ioConfigChanged() new output samplingRate %d, format %d channels %d frameCount %d latency %d",
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outputDesc->samplingRate, outputDesc->format, outputDesc->channels, outputDesc->frameCount, outputDesc->latency);
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} break;
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case OUTPUT_CLOSED: {
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if (gOutputs.indexOfKey(ioHandle) < 0) {
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LOGW("ioConfigChanged() closing unknow output! %d", ioHandle);
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break;
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}
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LOGV("ioConfigChanged() output %d closed", ioHandle);
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gOutputs.removeItem(ioHandle);
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for (int i = gStreamOutputMap.size() - 1; i >= 0 ; i--) {
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if (gStreamOutputMap.valueAt(i) == ioHandle) {
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gStreamOutputMap.removeItemsAt(i);
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}
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}
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} break;
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case OUTPUT_CONFIG_CHANGED: {
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int index = gOutputs.indexOfKey(ioHandle);
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if (index < 0) {
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LOGW("ioConfigChanged() modifying unknow output! %d", ioHandle);
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break;
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}
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if (param2 == 0) break;
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desc = (OutputDescriptor *)param2;
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LOGV("ioConfigChanged() new config for output %d samplingRate %d, format %d channels %d frameCount %d latency %d",
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ioHandle, desc->samplingRate, desc->format,
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desc->channels, desc->frameCount, desc->latency);
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OutputDescriptor *outputDesc = gOutputs.valueAt(index);
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delete outputDesc;
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outputDesc = new OutputDescriptor(*desc);
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gOutputs.replaceValueFor(ioHandle, outputDesc);
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} break;
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case INPUT_OPENED:
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case INPUT_CLOSED:
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case INPUT_CONFIG_CHANGED:
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break;
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}
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}
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void AudioSystem::setErrorCallback(audio_error_callback cb) {
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Mutex::Autolock _l(gLock);
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gAudioErrorCallback = cb;
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}
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bool AudioSystem::routedToA2dpOutput(int streamType) {
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switch(streamType) {
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case MUSIC:
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case VOICE_CALL:
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case BLUETOOTH_SCO:
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case SYSTEM:
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return true;
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default:
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return false;
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}
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}
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// client singleton for AudioPolicyService binder interface
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sp<IAudioPolicyService> AudioSystem::gAudioPolicyService;
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sp<AudioSystem::AudioPolicyServiceClient> AudioSystem::gAudioPolicyServiceClient;
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// establish binder interface to AudioFlinger service
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const sp<IAudioPolicyService>& AudioSystem::get_audio_policy_service()
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{
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gLock.lock();
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if (gAudioPolicyService.get() == 0) {
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sp<IServiceManager> sm = defaultServiceManager();
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sp<IBinder> binder;
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do {
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binder = sm->getService(String16("media.audio_policy"));
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if (binder != 0)
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break;
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LOGW("AudioPolicyService not published, waiting...");
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usleep(500000); // 0.5 s
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} while(true);
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if (gAudioPolicyServiceClient == NULL) {
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gAudioPolicyServiceClient = new AudioPolicyServiceClient();
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}
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binder->linkToDeath(gAudioPolicyServiceClient);
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gAudioPolicyService = interface_cast<IAudioPolicyService>(binder);
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gLock.unlock();
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} else {
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gLock.unlock();
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}
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return gAudioPolicyService;
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}
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status_t AudioSystem::setDeviceConnectionState(audio_devices device,
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device_connection_state state,
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const char *device_address)
|
|
{
|
|
const sp<IAudioPolicyService>& aps = AudioSystem::get_audio_policy_service();
|
|
if (aps == 0) return PERMISSION_DENIED;
|
|
|
|
return aps->setDeviceConnectionState(device, state, device_address);
|
|
}
|
|
|
|
AudioSystem::device_connection_state AudioSystem::getDeviceConnectionState(audio_devices device,
|
|
const char *device_address)
|
|
{
|
|
const sp<IAudioPolicyService>& aps = AudioSystem::get_audio_policy_service();
|
|
if (aps == 0) return DEVICE_STATE_UNAVAILABLE;
|
|
|
|
return aps->getDeviceConnectionState(device, device_address);
|
|
}
|
|
|
|
status_t AudioSystem::setPhoneState(int state)
|
|
{
|
|
const sp<IAudioPolicyService>& aps = AudioSystem::get_audio_policy_service();
|
|
if (aps == 0) return PERMISSION_DENIED;
|
|
|
|
return aps->setPhoneState(state);
|
|
}
|
|
|
|
status_t AudioSystem::setRingerMode(uint32_t mode, uint32_t mask)
|
|
{
|
|
const sp<IAudioPolicyService>& aps = AudioSystem::get_audio_policy_service();
|
|
if (aps == 0) return PERMISSION_DENIED;
|
|
return aps->setRingerMode(mode, mask);
|
|
}
|
|
|
|
status_t AudioSystem::setForceUse(force_use usage, forced_config config)
|
|
{
|
|
const sp<IAudioPolicyService>& aps = AudioSystem::get_audio_policy_service();
|
|
if (aps == 0) return PERMISSION_DENIED;
|
|
return aps->setForceUse(usage, config);
|
|
}
|
|
|
|
AudioSystem::forced_config AudioSystem::getForceUse(force_use usage)
|
|
{
|
|
const sp<IAudioPolicyService>& aps = AudioSystem::get_audio_policy_service();
|
|
if (aps == 0) return FORCE_NONE;
|
|
return aps->getForceUse(usage);
|
|
}
|
|
|
|
|
|
audio_io_handle_t AudioSystem::getOutput(stream_type stream,
|
|
uint32_t samplingRate,
|
|
uint32_t format,
|
|
uint32_t channels,
|
|
output_flags flags)
|
|
{
|
|
audio_io_handle_t output = 0;
|
|
if ((flags & AudioSystem::OUTPUT_FLAG_DIRECT) == 0) {
|
|
Mutex::Autolock _l(gLock);
|
|
output = AudioSystem::gStreamOutputMap.valueFor(stream);
|
|
LOGV_IF((output != 0), "getOutput() read %d from cache for stream %d", output, stream);
|
|
}
|
|
if (output == 0) {
|
|
const sp<IAudioPolicyService>& aps = AudioSystem::get_audio_policy_service();
|
|
if (aps == 0) return 0;
|
|
output = aps->getOutput(stream, samplingRate, format, channels, flags);
|
|
if ((flags & AudioSystem::OUTPUT_FLAG_DIRECT) == 0) {
|
|
Mutex::Autolock _l(gLock);
|
|
AudioSystem::gStreamOutputMap.add(stream, output);
|
|
}
|
|
}
|
|
return output;
|
|
}
|
|
|
|
status_t AudioSystem::startOutput(audio_io_handle_t output, AudioSystem::stream_type stream)
|
|
{
|
|
const sp<IAudioPolicyService>& aps = AudioSystem::get_audio_policy_service();
|
|
if (aps == 0) return PERMISSION_DENIED;
|
|
return aps->startOutput(output, stream);
|
|
}
|
|
|
|
status_t AudioSystem::stopOutput(audio_io_handle_t output, AudioSystem::stream_type stream)
|
|
{
|
|
const sp<IAudioPolicyService>& aps = AudioSystem::get_audio_policy_service();
|
|
if (aps == 0) return PERMISSION_DENIED;
|
|
return aps->stopOutput(output, stream);
|
|
}
|
|
|
|
void AudioSystem::releaseOutput(audio_io_handle_t output)
|
|
{
|
|
const sp<IAudioPolicyService>& aps = AudioSystem::get_audio_policy_service();
|
|
if (aps == 0) return;
|
|
aps->releaseOutput(output);
|
|
}
|
|
|
|
audio_io_handle_t AudioSystem::getInput(int inputSource,
|
|
uint32_t samplingRate,
|
|
uint32_t format,
|
|
uint32_t channels,
|
|
audio_in_acoustics acoustics)
|
|
{
|
|
const sp<IAudioPolicyService>& aps = AudioSystem::get_audio_policy_service();
|
|
if (aps == 0) return 0;
|
|
return aps->getInput(inputSource, samplingRate, format, channels, acoustics);
|
|
}
|
|
|
|
status_t AudioSystem::startInput(audio_io_handle_t input)
|
|
{
|
|
const sp<IAudioPolicyService>& aps = AudioSystem::get_audio_policy_service();
|
|
if (aps == 0) return PERMISSION_DENIED;
|
|
return aps->startInput(input);
|
|
}
|
|
|
|
status_t AudioSystem::stopInput(audio_io_handle_t input)
|
|
{
|
|
const sp<IAudioPolicyService>& aps = AudioSystem::get_audio_policy_service();
|
|
if (aps == 0) return PERMISSION_DENIED;
|
|
return aps->stopInput(input);
|
|
}
|
|
|
|
void AudioSystem::releaseInput(audio_io_handle_t input)
|
|
{
|
|
const sp<IAudioPolicyService>& aps = AudioSystem::get_audio_policy_service();
|
|
if (aps == 0) return;
|
|
aps->releaseInput(input);
|
|
}
|
|
|
|
status_t AudioSystem::initStreamVolume(stream_type stream,
|
|
int indexMin,
|
|
int indexMax)
|
|
{
|
|
const sp<IAudioPolicyService>& aps = AudioSystem::get_audio_policy_service();
|
|
if (aps == 0) return PERMISSION_DENIED;
|
|
return aps->initStreamVolume(stream, indexMin, indexMax);
|
|
}
|
|
|
|
status_t AudioSystem::setStreamVolumeIndex(stream_type stream, int index)
|
|
{
|
|
const sp<IAudioPolicyService>& aps = AudioSystem::get_audio_policy_service();
|
|
if (aps == 0) return PERMISSION_DENIED;
|
|
return aps->setStreamVolumeIndex(stream, index);
|
|
}
|
|
|
|
status_t AudioSystem::getStreamVolumeIndex(stream_type stream, int *index)
|
|
{
|
|
const sp<IAudioPolicyService>& aps = AudioSystem::get_audio_policy_service();
|
|
if (aps == 0) return PERMISSION_DENIED;
|
|
return aps->getStreamVolumeIndex(stream, index);
|
|
}
|
|
|
|
// ---------------------------------------------------------------------------
|
|
|
|
void AudioSystem::AudioPolicyServiceClient::binderDied(const wp<IBinder>& who) {
|
|
Mutex::Autolock _l(AudioSystem::gLock);
|
|
AudioSystem::gAudioPolicyService.clear();
|
|
|
|
LOGW("AudioPolicyService server died!");
|
|
}
|
|
|
|
// ---------------------------------------------------------------------------
|
|
|
|
|
|
// use emulated popcount optimization
|
|
// http://www.df.lth.se/~john_e/gems/gem002d.html
|
|
uint32_t AudioSystem::popCount(uint32_t u)
|
|
{
|
|
u = ((u&0x55555555) + ((u>>1)&0x55555555));
|
|
u = ((u&0x33333333) + ((u>>2)&0x33333333));
|
|
u = ((u&0x0f0f0f0f) + ((u>>4)&0x0f0f0f0f));
|
|
u = ((u&0x00ff00ff) + ((u>>8)&0x00ff00ff));
|
|
u = ( u&0x0000ffff) + (u>>16);
|
|
return u;
|
|
}
|
|
|
|
bool AudioSystem::isOutputDevice(audio_devices device)
|
|
{
|
|
if ((popCount(device) == 1 ) &&
|
|
((device & ~AudioSystem::DEVICE_OUT_ALL) == 0)) {
|
|
return true;
|
|
} else {
|
|
return false;
|
|
}
|
|
}
|
|
|
|
bool AudioSystem::isInputDevice(audio_devices device)
|
|
{
|
|
if ((popCount(device) == 1 ) &&
|
|
((device & ~AudioSystem::DEVICE_IN_ALL) == 0)) {
|
|
return true;
|
|
} else {
|
|
return false;
|
|
}
|
|
}
|
|
|
|
bool AudioSystem::isA2dpDevice(audio_devices device)
|
|
{
|
|
if ((popCount(device) == 1 ) &&
|
|
(device & (AudioSystem::DEVICE_OUT_BLUETOOTH_A2DP |
|
|
AudioSystem::DEVICE_OUT_BLUETOOTH_A2DP_HEADPHONES |
|
|
AudioSystem::DEVICE_OUT_BLUETOOTH_A2DP_SPEAKER))) {
|
|
return true;
|
|
} else {
|
|
return false;
|
|
}
|
|
}
|
|
|
|
bool AudioSystem::isBluetoothScoDevice(audio_devices device)
|
|
{
|
|
if ((popCount(device) == 1 ) &&
|
|
(device & (AudioSystem::DEVICE_OUT_BLUETOOTH_SCO |
|
|
AudioSystem::DEVICE_OUT_BLUETOOTH_SCO_HEADSET |
|
|
AudioSystem::DEVICE_OUT_BLUETOOTH_SCO_CARKIT))) {
|
|
return true;
|
|
} else {
|
|
return false;
|
|
}
|
|
}
|
|
|
|
bool AudioSystem::isLowVisibility(stream_type stream)
|
|
{
|
|
if (stream == AudioSystem::SYSTEM ||
|
|
stream == AudioSystem::NOTIFICATION ||
|
|
stream == AudioSystem::RING) {
|
|
return true;
|
|
} else {
|
|
return false;
|
|
}
|
|
}
|
|
|
|
bool AudioSystem::isInputChannel(uint32_t channel)
|
|
{
|
|
if ((channel & ~AudioSystem::CHANNEL_IN_ALL) == 0) {
|
|
return true;
|
|
} else {
|
|
return false;
|
|
}
|
|
}
|
|
|
|
bool AudioSystem::isOutputChannel(uint32_t channel)
|
|
{
|
|
if ((channel & ~AudioSystem::CHANNEL_OUT_ALL) == 0) {
|
|
return true;
|
|
} else {
|
|
return false;
|
|
}
|
|
}
|
|
|
|
bool AudioSystem::isValidFormat(uint32_t format)
|
|
{
|
|
switch (format & MAIN_FORMAT_MASK) {
|
|
case PCM:
|
|
case MP3:
|
|
case AMR_NB:
|
|
case AMR_WB:
|
|
case AAC:
|
|
case HE_AAC_V1:
|
|
case HE_AAC_V2:
|
|
case VORBIS:
|
|
return true;
|
|
default:
|
|
return false;
|
|
}
|
|
}
|
|
|
|
bool AudioSystem::isLinearPCM(uint32_t format)
|
|
{
|
|
switch (format) {
|
|
case PCM_16_BIT:
|
|
case PCM_8_BIT:
|
|
return true;
|
|
default:
|
|
return false;
|
|
}
|
|
}
|
|
|
|
//------------------------- AudioParameter class implementation ---------------
|
|
|
|
const char *AudioParameter::keyRouting = "routing";
|
|
const char *AudioParameter::keySamplingRate = "sampling_rate";
|
|
const char *AudioParameter::keyFormat = "format";
|
|
const char *AudioParameter::keyChannels = "channels";
|
|
const char *AudioParameter::keyFrameCount = "frame_count";
|
|
|
|
AudioParameter::AudioParameter(const String8& keyValuePairs)
|
|
{
|
|
char *str = new char[keyValuePairs.length()+1];
|
|
mKeyValuePairs = keyValuePairs;
|
|
|
|
strcpy(str, keyValuePairs.string());
|
|
char *pair = strtok(str, ";");
|
|
while (pair != NULL) {
|
|
if (strlen(pair) != 0) {
|
|
size_t eqIdx = strcspn(pair, "=");
|
|
String8 key = String8(pair, eqIdx);
|
|
String8 value;
|
|
if (eqIdx == strlen(pair)) {
|
|
value = String8("");
|
|
} else {
|
|
value = String8(pair + eqIdx + 1);
|
|
}
|
|
if (mParameters.indexOfKey(key) < 0) {
|
|
mParameters.add(key, value);
|
|
} else {
|
|
mParameters.replaceValueFor(key, value);
|
|
}
|
|
} else {
|
|
LOGV("AudioParameter() cstor empty key value pair");
|
|
}
|
|
pair = strtok(NULL, ";");
|
|
}
|
|
|
|
delete[] str;
|
|
}
|
|
|
|
AudioParameter::~AudioParameter()
|
|
{
|
|
mParameters.clear();
|
|
}
|
|
|
|
String8 AudioParameter::toString()
|
|
{
|
|
String8 str = String8("");
|
|
|
|
size_t size = mParameters.size();
|
|
for (size_t i = 0; i < size; i++) {
|
|
str += mParameters.keyAt(i);
|
|
str += "=";
|
|
str += mParameters.valueAt(i);
|
|
if (i < (size - 1)) str += ";";
|
|
}
|
|
return str;
|
|
}
|
|
|
|
status_t AudioParameter::add(const String8& key, const String8& value)
|
|
{
|
|
if (mParameters.indexOfKey(key) < 0) {
|
|
mParameters.add(key, value);
|
|
return NO_ERROR;
|
|
} else {
|
|
mParameters.replaceValueFor(key, value);
|
|
return ALREADY_EXISTS;
|
|
}
|
|
}
|
|
|
|
status_t AudioParameter::addInt(const String8& key, const int value)
|
|
{
|
|
char str[12];
|
|
if (snprintf(str, 12, "%d", value) > 0) {
|
|
String8 str8 = String8(str);
|
|
return add(key, str8);
|
|
} else {
|
|
return BAD_VALUE;
|
|
}
|
|
}
|
|
|
|
status_t AudioParameter::addFloat(const String8& key, const float value)
|
|
{
|
|
char str[23];
|
|
if (snprintf(str, 23, "%.10f", value) > 0) {
|
|
String8 str8 = String8(str);
|
|
return add(key, str8);
|
|
} else {
|
|
return BAD_VALUE;
|
|
}
|
|
}
|
|
|
|
status_t AudioParameter::remove(const String8& key)
|
|
{
|
|
if (mParameters.indexOfKey(key) >= 0) {
|
|
mParameters.removeItem(key);
|
|
return NO_ERROR;
|
|
} else {
|
|
return BAD_VALUE;
|
|
}
|
|
}
|
|
|
|
status_t AudioParameter::get(const String8& key, String8& value)
|
|
{
|
|
if (mParameters.indexOfKey(key) >= 0) {
|
|
value = mParameters.valueFor(key);
|
|
return NO_ERROR;
|
|
} else {
|
|
return BAD_VALUE;
|
|
}
|
|
}
|
|
|
|
status_t AudioParameter::getInt(const String8& key, int& value)
|
|
{
|
|
String8 str8;
|
|
status_t result = get(key, str8);
|
|
value = 0;
|
|
if (result == NO_ERROR) {
|
|
int val;
|
|
if (sscanf(str8.string(), "%d", &val) == 1) {
|
|
value = val;
|
|
} else {
|
|
result = INVALID_OPERATION;
|
|
}
|
|
}
|
|
return result;
|
|
}
|
|
|
|
status_t AudioParameter::getFloat(const String8& key, float& value)
|
|
{
|
|
String8 str8;
|
|
status_t result = get(key, str8);
|
|
value = 0;
|
|
if (result == NO_ERROR) {
|
|
float val;
|
|
if (sscanf(str8.string(), "%f", &val) == 1) {
|
|
value = val;
|
|
} else {
|
|
result = INVALID_OPERATION;
|
|
}
|
|
}
|
|
return result;
|
|
}
|
|
|
|
status_t AudioParameter::getAt(size_t index, String8& key, String8& value)
|
|
{
|
|
if (mParameters.size() > index) {
|
|
key = mParameters.keyAt(index);
|
|
value = mParameters.valueAt(index);
|
|
return NO_ERROR;
|
|
} else {
|
|
return BAD_VALUE;
|
|
}
|
|
}
|
|
}; // namespace android
|
|
|